MPEG-4 Bandwidth Prediction for Broadband Cable Networks

In this paper, we analyze the bandwidth requirements of MPEG-4 and propose and evaluate the accuracy of two new MPEG-4 video traffic models. The first predicts the size of the B-frame while the second predicts the size of the group of pictures (GOP). In a broadband cable network, the B-frame prediction model can be employed by a cable headend to provision video bandwidth efficiently or more importantly, reduce bit rate variability and bandwidth requirements via selective B-frame dropping, thereby minimizing buffering requirements and packet losses at the set top box. It will be shown that the model provides highly accurate prediction for a wide variety of video types, in particular, high action and regular movies encoded in high quality resolution. This is an important benefit since high motion MPEG videos demand the highest bandwidth and generate the widest range of bit rate requirements. Although the B-frame prediction model is very simple to implement, this paper also proposes two methods to simplify its operation further without sacrificing prediction accuracy. We then enhance the B-frame and GOP prediction models using a new scene change detector metric.      

MPEG4视频流量预测自适应算法的研究

为了保证用户的服务质量(QOS),宽带分组网在传送视频信息时需要进行动态带宽分配,而视频流量预测在动态带宽分配中发挥着重要的作用。本文从自相关性、自相似性的Hurst参数两个方面,阐明GOP时间尺度上的流量能够体现原始帧序列的流量特性,并在固定步长的LMS自适应算法(FSSA)的基础上提出的一种新的可变步长自适应算法(VSSA),在GOP的大时间尺度上预测MPEG4视频流量,通过大量的仿真实验表明,VSSA算法可以明显地改善预测性能。     

Supporting multi-user real-time VBR video service using dynamic bandwidth management based on prediction

Real-time variable bit rate (VBR) video is expected to take a significant portion of multimedia applications. However, plentiful challenges to VBR video service provision have been raised for its characteristic of high traffic abruptness. To support multi-user real-time VBR video transmission with high bandwidth utilization and satisfied quality of service (QoS), this article proposes a practical dynamic bandwidth management scheme. This scheme forecasts future media rate of VBR video by employing time-domain adaptive linear predictor and using media delivery index (MDI) as both QoS measurement and complementary management reference. In addition, to support multi-user application, an adjustment priorities classified strategy is also put forward. Finally, a test-bed based on this management scheme is established. The experimental results demonstrate that the scheme proposed in this article is efficient with bandwidth utilization increased by 20%-60% compared to a fixed service rate and QoS guaranteed.     

Efficient traffic prediction scheme for real-time VBR MPEG videotransmission over high-speed networks

The efficient transportation of real-time variable bit rate (VBR) video traffic in high-speed networks is currently an active area of research. The capability to predict VBR video traffic can significantly improve the effectiveness of numerous management tasks, including dynamic bandwidth allocation and congestion control. This paper proposes an adaptive traffic prediction method for VBR MPEG videos, a major multimedia application. Rapid traffic variations due to scene changes are analyzed, then a prediction scheme using the identification of scene changes related to I and P frames is presented. For predicting multiplexed MPEG traffic, a prediction interval is derived that represents a highly correlated traffic sequence. In addition, to reduce the prediction error, a less fluctuating signal instead of the original multiplexed traffic is used as the input for the predictor. Simulation results show that the proposed method is able to predict the original traffic more accurately than the conventional LMS method     

Video traffic prediction using source information for renegotiatedCBR

A prediction method for variable bit rate video traffic based on source information is presented. By analysing the source before encoding, any changes in the source traffic are detected and the bit rate of the current frame is predicted. In addition, a new channel rate for renegotiated constant bit rate service can be determined for future frames using the predicted bit rate and a bit rate updating algorithm. Simulation results show that the proposed method achieves a better prediction and requires a smaller buffer than the conventional approach     

Feedback control mechanisms for real-time multipoint video services

While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include: (1) implosion of feedback returning to the source as the number of multicast destinations increases and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. A new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multilayered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a constant bit rate (CBR) connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level     

Scene oriented model for VBR video

A new model for variable bit rate (VBR) video traffic is presented. The model, which could be used as a traffic generator, considers two time scales: scenes, for periods of several minutes, and groups of pictures (GOP), for periods of half a second. To model the scene changes, a Markov chain is used. For the GOP level. A modification for the projected autoregressive (PAR) model is proposed so that the fitting of the autocorrelation function is improved. The model is used to imitate two real MPEG video signals, showing that it is able to accurately capture the behaviour of the real traffic in a queue     

Wyner-Ziv视频编码中无反馈速率控制算法研究

为了避免在分布式视频编码系统中使用反馈信道,提出了一种基于Wyner-Ziv编码的无反馈速率控制算法。首先,利用目标码率和目标帧率进行GOP层码率分配;然后,根据原始图像的帧间相关性动态选择量化因子和量化矩阵来分配每个GOP内关键帧和Wyner-Ziv帧的比特数;接下来,利用系数带级的相关性计算相关噪声模型参数,并选择对应的LDPC校验矩阵,提出Wyner-Ziv帧的无反馈比特面速率控制算法。实验结果表明,在给定目标码率下,所提算法的编码码率误差小于0.57%,且与现有无反馈速率控制算法相比,解码恢复图像的PSNR(峰值信噪比)可以提高1dB。另外,该算法基本没有增加编码端复杂度,可用于实际分布式视频通信系统。     

An analysis of statistical multiplexing in an ATM transport network

The traffic characteristics of an asynchronous transfer mode (ATM) network are analyzed by theoretical methods. A new method is proposed to express the burstiness of the cell arrival process. Both statistical multiplexing and statistical bandwidth allocation are quantitatively evaluated. When packetized video traffic and voice traffic are multiplexed, the number of multiplexable sources strongly depends on the peak bit rate of the multiplexed video sources, and statistical bandwidth allocation is ineffective without control. On the other hand, lowering the peak bit rate of video traffic effectively improves bandwidth utilization     

Efficient channel utilization for real-time video in OVSF-CDMA systems with QoS assurance

In this paper, the utilization of real-time video service in the downlink of an orthogonal variable spreading factor code division multiple access (OVSF-CDMA) system is studied. By modeling the video traffic and wireless channel as a joint Markov modulated process, and properly partitioning the states of the Markov process, an adaptive rate allocation scheme is proposed for real-time video transmission with quality of service provisioning while achieving high channel utilization. The scheme is applicable for packet switching and frame-by-frame real-time video transmission, and incorporates both the physical layer and network layer characteristics. For QoS provisions, the closed form expressions of packet delay and loss probability are derived based on the Markov model. Analytical and simulation results demonstrate that the proposed scheme can significantly improve the channel utilization over the commonly used effective bandwidth approach.     

一种平滑的可变步长算法

本文从长时相关性(LRD)和突发性两个方面,阐明图像组(GOP)时间尺度上的流量能够体现原有视频帧序列的流量特性。并在固定步长自适应算法(FSSA)和可变步长自适应算法(VSSA)的基础上提出了一种新的平滑的可变步长自适应算法(S-VSSA),这种算法可以在GOP所表示的大时间尺度上预测MPEG-4视频流量。仿真实验结果显示:该算法收敛速度快,预测误差低,可以明显改善预测性能。     

Short‐term MPEG‐4 video traffic prediction using ANFIS

Multimedia traffic and particularly MPEG‐coded video streams are growing to be a major traffic component in high‐speed networks. Accurate prediction of such traffic enhances the reliable operation and the quality of service of these networks through a more effective bandwidth allocation and better control strategies. However, MPEG video traffic is characterized by a periodic correlation structure, a highly complex bit rate distribution and very noisy streams. Therefore, it is considered an intractable problem. This paper presents a neuro‐fuzzy short‐term predictor for MPEG‐4‐coded videos. The predictor is based on the Adaptive Network Fuzzy Inference System (ANFIS) to perform single‐step predictions for the I, P and B frames. Short‐term predictions are also examined using smoothed signals of the video sequences. The ANFIS prediction results are evaluated using long entertainment and broadcast video sequences and compared to those obtained using a linear predictor. ANFIS is capable of providing accurate prediction and has the added advantage of being simple to design and to implement. Copyright © 2005 John Wiley & Sons, Ltd.     

Fair Delay Optimization-based Resource Allocation Algorithm for Video Traffic over Wireless Multimedia System

The major issue related to the realization of wireless multimedia system is the design of suitable medium access control (MAC) protocol. The design challenge is to maximize the utilization of the limited wireless resources while guaranteeing the various quality of service requirements for all traffic classes especially for the stringent real-time constraint of real time variable bit rate (rt-VBR) video service. In this paper a novel resource allocation algorithm for video traffic is proposed. The proposed allocation algorithm aims to provide fair delay for video packets by minimizing the delay difference among transmitted video packets. At the same time it adaptively controls the allocated resources (bandwidth) for video traffic around the corresponding average bit rate, and has the ability of controlling the quality of service (QoS) offered for video traffic in terms of packet loss probability and average delay. A minimized control overhead of only two bits is needed to increase the utilization efficiency. Simulation results show that the proposed algorithm achieves very high utilization and provides nearly fair delay among video packets. Its efficiency is also investigated under traffic integration condition with voice and data traffic to show that the QoS offered to video traffic does not change in the presence of the highest priority voice traffic while data traffic increases the channel utilization to 98% by using the remaining bandwidth after voice and video traffic while a good QoS is offered to voice and data traffic.

QoS support for integrated services over CATV

Cable TV has emerged as a promising access network infrastructure for the delivery of voice, video, and high-speed data traffic. A central issue in the design of protocols for CATV networks is to support different levels of QoS for diverse user applications. While CATV service providers and equipment have standardized, in the so-called MCNS protocol, the basic network architecture and interfaces, issues in the MAC layer for QoS support are likely to be left for differentiation in vendor products. This article first presents an overview of the basic CATV network architectural assumptions and the set of QoS requirements for supporting integrated services over CATV. It then discusses a MAC layer scheduling protocol that can efficiently multiplex constant bit rate traffic, such as voice over IP with guaranteed delay bound, and best-effort traffic, such as data services with minimum bit rate guarantee, while achieving fairness on any excess available bandwidth. The performance of this algorithm is illustrated by simulation results using Opnet. We also discuss a dynamic polling mechanism that enhances the link utilization while preserving delay bounds for latency-critical traffic     

Design of frame dependency for VCR streaming videos

The structure of Group of pictures (GOP) is favorable for GOP-based random-access operations, while inside the GOP, the sequential coding dependency between frames is unfavorable for frame-based random-access or fast-scan operations. In a video streaming system, when some VCR operations such as fast-scan operations are performed, they may cause serious server load or network load due to the speed-up of transmitting frames. In this paper, we formulated the fast-scan and random-access operations of VCR functionality, and proposed a novel frame dependency to eliminate extra server load and minimize requirements of the network bandwidth when performing fast-scan and random-access operations. With the proposed structure, the server does not have to speed up the transmission rate of frames to fulfill the requested playback rate. The number of frames to be transmitted is also significantly reduced, and thus the network load can be decreased.     

Efficient Traffic Scheduling for Real Time VBR MPEG Video Transmission Over DOCSIS-Based HFC Networks

Data-over-cable service interface specifications (DOCSIS), the de facto standard in the cable industry, defines a scheduling service called real-time polling service (rtPS) to provision quality of service (QoS) transmission of real-time variable bit rate (VBR) videos. However, the rtPS service intrinsically has high latency, which makes it not applicable to real-time traffic transport. In this paper, we present a novel traffic scheduling algorithm for hybrid fiber coax (HFC) networks based on DOCSIS that aims to provide QoS for real-time VBR video transmissions. The novel characteristics of this algorithm, as compared to those described in published literatures, include 1) it predicts the bandwidth requirements for future traffic using a novel traffic predictor designed to provide simple yet accurate online prediction; and 2) it takes the attributes of physical (PHY) layer, media access control (MAC) layer and application layer into consideration. In addition, the proposed traffic scheduling algorithm is completely compatible with the DOCSIS specification and does not require any protocol changes. We analyze the performance of the proposed traffic predictor and traffic scheduling algorithm using real-life MPEG video traces. Simulation results indicate that 1) the proposed traffic predictor significantly outperforms previously published techniques with respect to the prediction error and 2) Compared with several existing scheduling algorithms, the proposed traffic scheduling algorithm surpasses other mechanisms in terms of channel utilization, buffer usage, packet delay, and packet loss rate.     

基于VBR视频流量预测技术的DVB-H复用器调度算法

DVB-H采用的时间分片技术对现有的复用器算法提出了挑战.本文在VBR视频流量预测技术的基础上提出了基于VBR视频流量预测技术的DVB-H网络复用器调度算法.本算法根据VBR视频码率的预测值,预先分配下一调度周期的带宽,满足了时间分片技术对复用器调度算法的要求.与CBR调度算法相比,本算法具有较小的时延和较高的带宽利用率.     

New queueing strategies employing service interval-based priority for real-time communications in atm networks

In this paper, we propose queueing strategies employing the service interval-based priority (sip) which can provide delay-bounded, and loss-free services, while maximizing bandwidth utilization in the atm network. We also describe a variation of the sip, the residual service interval-based priority (rsip) which can achieve almost full utilization by assigning priorities dynamically on the basis of the residual service interval. We store the realtime cells belonging to different connections in logically separated queues, and for each queue, we set a parameter called service interval, during which only one cell is allowed to be transmitted. The sip server takes and transmits the head-of-line (hol) cell of the queue which has the smallest service interval, while the rsip server selects the queue with the smallest residual service interval. When there is no eligible real-time cell, it transmits non-real-time cell, thus enabling a maximized bandwidth utilization. Employing the above queueing strategies, we analyze the delay characteristics deterministically with the leaky bucket bounded input traffic and then dimension the optimal service interval. In dimensioning the service interval and buffer space of each real-time service queue, we consider burstiness of traffic in conjunction with delay constraints, so that bandwidth utilization can get maximized. In addition, we consider the issues of protection from malicious users, average bandwidth utilization, and coupling between the delay bound and the bandwidth allocation granularity.     

Video traffic prediction based on source information and preventive channel rate decision for RCBR

In this paper, we address the problem of dynamic bandwidth allocation in real-time video transmission. Firstly, a source traffic prediction method is proposed which is based on the rate-distortion relation of source video. This method can detect changes in the source traffic level before encoding by using source information. Secondly, a preventive channel rate decision algorithm, called PCRD, is proposed. The transmission rate bounds are derived from the constraints of the encoder and decoder buffers based on the predicted bit-rate of video frames. From simulation results, the proposed traffic prediction method is shown effective in detecting scene changes and estimating changed traffic levels. Also, the PCRD method is shown to have low renegotiation cost and high channel utilization without violating delay constraints.